The present disclosure relates generally to facsimile transmission compensation, and more particularly to adaptive facsimile transmission redundancy that can compensate for problematic issues encountered in facsimile transmission.
Facsimile document transmission continues to have an important role in business communications for a number of reasons, including the ability to transfer images not stored on a local computer, legal acceptance of handwritten signatures, realtime confirmation of receipt, confidence in what has been sent/received, and the ability to provide a ‘tamper resistant’ copy of the information transferred. Since facsimile-enabled devices can be used with existing telecommunications networks, their popularity has increased to the point where such devices enjoy the advantage of being ubiquitous and familiar to users on a global scale. Such devices also may be shared by a number of individuals, such as in a workplace environment, reducing costs while still allowing for the sending and receiving of documents to be relatively efficient among a group of persons.
While facsimile communications have previously been implemented over circuit-switched networks, such as the public switched telephone network (PSTN), packet-switched networks, such as Internet Protocol (IP) networks, have been implemented to carry communications including facsimile communications. As these different types of networks continue to coexist, translation and communication between them has become (and should continue to be) an important part of communications, including facsimile communications.
Translation between circuit-switched and packet-switched communication networks typically involves the use of translation between different protocols, and is often performed by gateways, sometimes referred to as IP media gateways. A gateway can carry different types of communications between various network types, such as an IP network and a PSTN network. Such different types of communications may include voice or facsimile, for example. The gateway typically provides protocol translation service between the networks for these different types of communications. Facsimile transmissions typically adhere to the International Telecommunication Union Telecommunications Standardization Sector (ITU-T) Recommendation T.30, and are often implemented using the realtime facsimile transmission specification under the ITU-T Recommendation T.38.
IP networks are inherently asynchronous, tend to have greater latency and are relatively ‘lossy’ (lose or drop packets) as compared to PSTN networks, which typically operate on a time-division multiplexed (TDM) basis. While these characteristics of IP networks are known to adversely impact realtime communications, including both voice over IP (VoIP) and facsimile over IP (FoIP) communications, the impact to facsimile communications is typically more pronounced. Various solutions have been provided to address drawbacks related to realtime IP network communications, however, such solutions have tended to be focussed on voice data and the characteristics of voice-based calls. Facsimile users thus have sometimes reported negative experiences when attempting to perform facsimile transmissions over packet-switched networks.
Facsimile transmissions over packet-switched networks, including IP networks, typically use a facsimile transport protocol such as the T.38 realtime facsimile standard or the G.711 RTP (realtime transport protocol) facsimile pass-through codec, hereinafter referred to as the “G.711” codec. Often, the G.711 codec is used for facsimile pass-through when the T.38 protocol is unavailable at a given network node. The G.711 codec is suited for facsimile transmission in packet-switched network due to the low compression levels involved in implementing the codec. VoIP implementations typically support the G.711 codec as a default for pass-through facsimile transmissions.
In general, a VoIP network is optimized for handling voice traffic, where the particular characteristics of voice transmissions can be used to tailor components and/or operations in the VoIP network to improve the speed and reduce the bandwidth of voice communications. The system design and compensation provided for voice communications in an IP network tend to prove problematic for facsimile transmissions for a number of reasons. For example, the asynchronous nature of the IP network can create issues for those facsimile transmissions that are implemented in realtime. IP networks also tend to experience packet loss, which may not significantly degrade voice communications, but nevertheless can cause a facsimile transmission to degrade significantly or fail. Moreover, the asynchronous nature of packet-switched networks can result in variable latency or propagation delays, sometimes referred to as “jitter.” The variable latency can be caused by different routes being taken through the network, as well as various conditions of network operation. Issues related to jitter are typically addressed with a jitter buffer that queues up packets of data at a network node to help alleviate the disjointedness of propagation delays and smooth the transmission of data packets. Jitter buffers tend to increase overall delay for transmissions through a packet-switched network, and are therefore usually designed to be relatively small in length/duration to avoid uncomfortably long pauses in voice communications, which may prompt both parties in a voice communication to attempt to speak at the same time. However, reduced size jitter buffers may lead to packet loss when propagation delays are significant in comparison to jitter buffer length and the reduced size jitter buffer may be unable to accommodate the propagation delay.
In view of the above difficulties encountered in implementing a packet-switched network that handles both realtime voice and facsimile transmissions, and with the bias for transmission tending toward providing compensation for voice communications, facsimile transmissions tend to be more significantly impacted by the above noted issues. For example, a voice transmission through an IP network may be able to tolerate 5% packet loss without significant degradation in voice quality in terms of a mean opinion score (MOS), which measures a human users view of the quality of a communication experience in a network. At the same time, a T.38 facsimile transmission tends to experience significant degradation with the same or less packet loss, while a G.711 transmission tends to experience significant degradation with packet losses less than those that might generally degrade a T.38 facsimile transmission.
With a communication network that has managed network access such as may include non-internet communications, most SIP trunk providers that manage such networks report a relatively low level of packet loss, for example, less than the above mentioned 5% packet loss. While this level of packet loss may be considered acceptable for the majority of voice communications, it still represents a significant degradation for facsimile transmission service, particularly for G.711 transmissions.
One conventional technique for addressing the above-noted issues for facsimile transmissions in a packet-switched network is to provide forward error correction. One form of forward error correction that can be readily implemented to correct for packet loss is to provide redundancy in T.38 or G.711 transmissions. The redundancy implementation for forward error correction is typically preferred over retransmission techniques, which tend to be less useful due to the timing constraints of realtime communications involving facsimile transmissions. With redundancy techniques to implement forward error correction, a transmitting node essentially duplicates a media payload in consecutive packets. One of the drawbacks in implementing redundancy is that a standard call, such as may be implemented with a G.711 transmission, consumes nearly twice the bandwidth of a call that is managed without redundancy. Another drawback of redundancy is the increased delay in communications, which sometimes can lead to degradation or failure of facsimile communications in conjunction with the propagation delay issues inherent in packet-switched networks. For example, delays accumulated over multiple network devices or components can cause facsimile failure if the round trip delay exceeds a T.30 facsimile transmission timeout value. Typically, due to the increased bandwidth requirements, service providers tend not to offer the option for enabling redundancy for facsimile transmissions, particularly for G.711 transmissions, due to the increased bandwidth usage and latency that occur regardless of whether the call is voice or facsimile, and regardless of network conditions.
It would be desirable to overcome the drawbacks related to facsimile transmissions in a communication network with relatively high latency. It would also be desirable to avoid facsimile transmission failure when using communication networks that experience packet loss without significantly increasing bandwidth usage.